Sirannon (streaming)
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Sirannon (streaming)
Sirannon is a free, open-source, media server and client. The goal is to aid in video research and experimental streaming. Sirannon allows the programmer to create a wide variety of media-handling components such as streaming, reading, writing, packetizing. By organizing these components in a workflow the programmer can create many applications such as a media server, media proxy or video tool. Sirannon was introduced at the ACM multimedia conference in October 2009 under its former name xStreamer. Sirannon has been accepted as part of the reference toolchain, defined in the final testplan of the Video Quality Experts Group (VQEG) Hybrid Perceptual/Bitstream project, for streaming video sequences and simulating network impairments. Features * Protocol transcoding – universal server – universal client The main feature is the combination of universal server (RTSP, HTTP, RTMP, RTMPT) and universal client (RTSP, HTTP, RTMP, RTMPT). This combination gives Sirannnon the ability ...
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Python (programming Language)
Python is a high-level programming language, high-level, general-purpose programming language. Its design philosophy emphasizes code readability with the use of significant indentation. Python is type system#DYNAMIC, dynamically type-checked and garbage collection (computer science), garbage-collected. It supports multiple programming paradigms, including structured programming, structured (particularly procedural programming, procedural), object-oriented and functional programming. It is often described as a "batteries included" language due to its comprehensive standard library. Guido van Rossum began working on Python in the late 1980s as a successor to the ABC (programming language), ABC programming language, and he first released it in 1991 as Python 0.9.0. Python 2.0 was released in 2000. Python 3.0, released in 2008, was a major revision not completely backward-compatible with earlier versions. Python 2.7.18, released in 2020, was the last release of ...
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MPEG-2
MPEG-2 (a.k.a. H.222/H.262 as was defined by the ITU) is a standard for "the generic coding of moving pictures and associated audio information". It describes a combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and H.265/HEVC, backwards compatibility with existing hardware and software means it is still widely used, for example in over-the-air digital television broadcasting and in the DVD-Video standard. Main characteristics MPEG-2 is widely used as the format of digital television signals that are broadcast by terrestrial (over-the-air), cable, and direct broadcast satellite TV systems. It also specifies the format of movies and other programs that are distributed on DVD and similar discs. TV stations, TV receivers, DVD players, and other equipment are ...
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User Datagram Protocol
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages (transported as datagrams in Network packet, packets) to other hosts on an Internet Protocol (IP) network. Within an IP network, UDP does not require prior communication to set up communication channels or data paths. UDP is a connectionless protocol, meaning that messages are sent without negotiating a connection and that UDP does not keep track of what it has sent. UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram. It has no Handshake (computing), handshaking dialogues and thus exposes the user's program to any Reliability (computer networking), unreliability of the underlying network; there is no guarantee of delivery, ordering, or duplicate protection. If error-correction facilities are needed at the network interface level, an applica ...
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Real-time Transport Protocol
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features. RTP typically runs over User Datagram Protocol (UDP). RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published i ...
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RTSP
The Real-Time Streaming Protocol (RTSP) is an application-level network protocol designed for multiplexing and packetizing multimedia transport streams (such as interactive media, video and audio) over a suitable transport protocol. RTSP is used in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between endpoints. Clients of media servers issue commands such as ''play'', ''record'' and ''pause'', to facilitate real-time control of the media streaming from the server to a client (video on demand) or from a client to the server (voice recording). History RTSP was developed by RealNetworks, Netscape and Columbia University. The first draft was submitted to IETF in October 1996 by Netscape and Progressive Networks, after which Henning Schulzrinne from Columbia University submitted "RTSP՚" ("RTSP prime") in December 1996. The two drafts were merged for standardization by the Multi ...
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RTMPT
Real-Time Messaging Protocol (RTMP) is a communication protocol for streaming audio, video, and data over the Internet. Originally developed as a proprietary protocol by Macromedia for streaming between Flash Player and the Flash Communication Server, Adobe (which acquired Macromedia) has released an incomplete version of the specification of the protocol for public use. The RTMP protocol has multiple variations: # RTMP proper, the "plain" protocol which works on top of Transmission Control Protocol (TCP) and uses port number 1935 by default. # RTMPS, which is RTMP over a Transport Layer Security (TLS/SSL) connection. # RTMPE, which is RTMP encrypted using Adobe's own security mechanism. While the details of the implementation are proprietary, the mechanism uses industry standard cryptographic primitives. # RTMPT, which is encapsulated within HTTP requests to traverse firewalls. RTMPT is frequently found utilizing cleartext requests on TCP ports 80 and 443 to bypass most corpora ...
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Real Time Messaging Protocol
Real-Time Messaging Protocol (RTMP) is a communication protocol for streaming media, streaming audio, video, and data over the Internet. Originally developed as a proprietary protocol by Macromedia for streaming between Flash Player and the Flash Communication Server, Adobe Systems, Adobe (which acquired Macromedia) has released an incomplete version of the specification of the protocol for public use. The RTMP protocol has multiple variations: # RTMP proper, the "plain" protocol which works on top of Transmission Control Protocol (TCP) and uses port number 1935 by default. # RTMPS, which is RTMP over a Transport Layer Security (TLS/SSL) connection. # RTMPE, which is RTMP encrypted using Adobe's own security mechanism. While the details of the implementation are proprietary, the mechanism uses industry standard cryptographic primitives. # RTMPT, which is encapsulation (networking), encapsulated within HTTP requests to traverse firewall (computing), firewalls. RTMPT is frequently f ...
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Matroska
Matroska (styled Matroška) is a project to create a container format that can hold an unlimited number of video, audio, picture, or subtitle tracks in one file. The Matroska Multimedia Container is similar in concept to other containers like AVI, MP4, or Advanced Systems Format (ASF), but is an open standard. Matroska file extensions are ''.mkv'' for video (which may include subtitles or audio), ''.mk3d'' for stereoscopic video, ''.mka'' for audio-only files (which may include subtitles), and ''.mks'' for subtitles only. History The project was announced on 6 December 2002 as a fork of the Multimedia Container Format (MCF), after disagreements between MCF lead developer Lasse Kärkkäinen and soon-to-be Matroska founder Steve Lhomme about the use of the Extensible Binary Meta Language (EBML) instead of a binary format. This coincided with a 6-month coding break by the MCF's lead developer for his military service, during which most of the community quickly migrated ...
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Audio Video Interleave
Audio Video Interleave (also Audio Video Interleaved and known by its initials and filename extension AVI, usually pronounced ) is a proprietary multimedia container format and Windows standard introduced by Microsoft in November 1992 as part of its Video for Windows software. AVI files can contain both audio and video data in an uncompressed file container that allows synchronous audio-with-video playback. Like the DVD video format, AVI files support multiple streaming audio and video, although these features are seldom used. Codecs popularly used for AVI include DivX and Xvid, although many other codecs can also be contained in an AVI file. Many AVI files use the file format extensions developed by the Matrox OpenDML group in February 1996. These files are supported by Microsoft, and are unofficially called AVI 2.0. In 2010 the US government's National Archives and Records Administration defined AVI as the official wrapper for preserving digital video. History ...
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MPEG Transport Stream
MPEG transport stream (MPEG-TS, MTS) or simply transport stream (TS) is a standard digital container format for transmission and storage of audio, video, and Program and System Information Protocol (PSIP) data. It is used in broadcast systems such as DVB, ATSC and IPTV. Transport stream specifies a container format encapsulating packetized elementary streams, with error correction and synchronization pattern features for maintaining transmission integrity when the communication channel carrying the stream is degraded. Transport streams differ from the similarly named MPEG program stream in several important ways: program streams are designed for reasonably reliable media, such as discs (like DVDs), while transport streams are designed for less reliable transmission, namely terrestrial or satellite broadcast. Further, a transport stream may carry multiple programs. Transport stream is specified in ''MPEG-2 Part 1, Systems'', formally known as '' ISO/IEC standard 1 ...
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MPEG Program Stream
Program stream (PS or MPEG-PS) is a container format (digital), container format for multiplexing digital audio, video and more. The PS format is specified in MPEG-1 Part 1 (ISO/IEC 11172-1) and MPEG-2 Part 1, Systems (ISO/IEC standard 13818-1/ITU-T H.222.0). The MPEG-2 Program Stream is analogous and similar to ISO/IEC 11172 Systems layer and it is forward compatible.ISO (2000-12-01ISO/IEC 13818-1 : 2000, Second editionPage X, Retrieved on 2009-07-25 Program streams are used on DVD-Video discs and HD DVD video discs, but with some restrictions and extensions.DVD - MPeg differences
Retrieved on 2009-07-24
The filename extensions are VOB and Enhanced VOB, EVO respectively.


Coding structure

Program streams are created by combining one or more Packetized Elementary Streams (PES), which have a common time ...
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AMR-WB
Adaptive Multi-Rate Wideband (AMR-WB) is a patented Wideband audio, wideband speech coding, speech audio coding standard developed based on Adaptive Multi-Rate audio codec, Adaptive Multi-Rate encoding, using a similar methodology to algebraic code-excited linear prediction (ACELP). AMR-WB provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders which in general are optimized for Plain old telephone service, POTS wireline quality of 300–3400 Hz. AMR-WB was developed by Nokia and VoiceAge and it was first specified by 3GPP. AMR-WB is codified as G.722.2, an ITU-T standard speech codec, formally known as ''Wideband coding of speech at around 16 kbit/s using Adaptive Multi-Rate Wideband (AMR-WB)''. G.722.2 AMR-WB is the same codec as the 3GPP AMR-WB. The corresponding 3GPP specifications are TS 26.190 for the speech codec and TS 26.194 for the Voice Activity Detector. The AMR-WB format has the following par ...
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